🛠️🐜 Antkeeper superbuild with dependencies included
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276 lines
5.9 KiB
276 lines
5.9 KiB
#ifndef CORE_VOICE_H
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#define CORE_VOICE_H
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#include <array>
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#include <atomic>
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#include <bitset>
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#include <memory>
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#include <stddef.h>
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#include <string>
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#include "albyte.h"
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#include "almalloc.h"
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#include "aloptional.h"
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#include "alspan.h"
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#include "bufferline.h"
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#include "buffer_storage.h"
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#include "devformat.h"
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#include "filters/biquad.h"
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#include "filters/nfc.h"
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#include "filters/splitter.h"
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#include "mixer/defs.h"
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#include "mixer/hrtfdefs.h"
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#include "resampler_limits.h"
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#include "uhjfilter.h"
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#include "vector.h"
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struct ContextBase;
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struct DeviceBase;
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struct EffectSlot;
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enum class DistanceModel : unsigned char;
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using uint = unsigned int;
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#define MAX_SENDS 6
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enum class SpatializeMode : unsigned char {
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Off,
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On,
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Auto
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};
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enum class DirectMode : unsigned char {
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Off,
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DropMismatch,
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RemixMismatch
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};
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/* Maximum number of extra source samples that may need to be loaded, for
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* resampling or conversion purposes.
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*/
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constexpr uint MaxPostVoiceLoad{MaxResamplerEdge + UhjDecoder::sFilterDelay};
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enum {
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AF_None = 0,
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AF_LowPass = 1,
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AF_HighPass = 2,
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AF_BandPass = AF_LowPass | AF_HighPass
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};
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struct DirectParams {
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BiquadFilter LowPass;
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BiquadFilter HighPass;
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NfcFilter NFCtrlFilter;
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struct {
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HrtfFilter Old;
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HrtfFilter Target;
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alignas(16) std::array<float,HrtfHistoryLength> History;
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} Hrtf;
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struct {
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std::array<float,MAX_OUTPUT_CHANNELS> Current;
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std::array<float,MAX_OUTPUT_CHANNELS> Target;
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} Gains;
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};
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struct SendParams {
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BiquadFilter LowPass;
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BiquadFilter HighPass;
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struct {
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std::array<float,MAX_OUTPUT_CHANNELS> Current;
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std::array<float,MAX_OUTPUT_CHANNELS> Target;
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} Gains;
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};
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struct VoiceBufferItem {
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std::atomic<VoiceBufferItem*> mNext{nullptr};
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CallbackType mCallback{nullptr};
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void *mUserData{nullptr};
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uint mSampleLen{0u};
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uint mLoopStart{0u};
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uint mLoopEnd{0u};
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al::byte *mSamples{nullptr};
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};
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struct VoiceProps {
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float Pitch;
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float Gain;
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float OuterGain;
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float MinGain;
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float MaxGain;
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float InnerAngle;
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float OuterAngle;
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float RefDistance;
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float MaxDistance;
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float RolloffFactor;
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std::array<float,3> Position;
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std::array<float,3> Velocity;
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std::array<float,3> Direction;
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std::array<float,3> OrientAt;
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std::array<float,3> OrientUp;
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bool HeadRelative;
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DistanceModel mDistanceModel;
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Resampler mResampler;
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DirectMode DirectChannels;
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SpatializeMode mSpatializeMode;
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bool DryGainHFAuto;
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bool WetGainAuto;
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bool WetGainHFAuto;
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float OuterGainHF;
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float AirAbsorptionFactor;
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float RoomRolloffFactor;
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float DopplerFactor;
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std::array<float,2> StereoPan;
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float Radius;
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float EnhWidth;
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/** Direct filter and auxiliary send info. */
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struct {
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float Gain;
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float GainHF;
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float HFReference;
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float GainLF;
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float LFReference;
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} Direct;
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struct SendData {
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EffectSlot *Slot;
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float Gain;
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float GainHF;
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float HFReference;
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float GainLF;
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float LFReference;
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} Send[MAX_SENDS];
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};
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struct VoicePropsItem : public VoiceProps {
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std::atomic<VoicePropsItem*> next{nullptr};
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DEF_NEWDEL(VoicePropsItem)
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};
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enum : uint {
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VoiceIsStatic,
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VoiceIsCallback,
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VoiceIsAmbisonic,
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VoiceCallbackStopped,
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VoiceIsFading,
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VoiceHasHrtf,
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VoiceHasNfc,
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VoiceFlagCount
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};
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struct Voice {
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enum State {
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Stopped,
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Playing,
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Stopping,
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Pending
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};
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std::atomic<VoicePropsItem*> mUpdate{nullptr};
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VoiceProps mProps;
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std::atomic<uint> mSourceID{0u};
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std::atomic<State> mPlayState{Stopped};
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std::atomic<bool> mPendingChange{false};
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/**
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* Source offset in samples, relative to the currently playing buffer, NOT
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* the whole queue.
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*/
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std::atomic<uint> mPosition;
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/** Fractional (fixed-point) offset to the next sample. */
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std::atomic<uint> mPositionFrac;
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/* Current buffer queue item being played. */
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std::atomic<VoiceBufferItem*> mCurrentBuffer;
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/* Buffer queue item to loop to at end of queue (will be NULL for non-
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* looping voices).
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*/
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std::atomic<VoiceBufferItem*> mLoopBuffer;
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/* Properties for the attached buffer(s). */
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FmtChannels mFmtChannels;
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FmtType mFmtType;
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uint mFrequency;
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uint mFrameStep; /**< In steps of the sample type size. */
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uint mFrameSize; /**< In bytes. */
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AmbiLayout mAmbiLayout;
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AmbiScaling mAmbiScaling;
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uint mAmbiOrder;
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std::unique_ptr<DecoderBase> mDecoder;
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uint mDecoderPadding{};
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/** Current target parameters used for mixing. */
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uint mStep{0};
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ResamplerFunc mResampler;
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InterpState mResampleState;
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std::bitset<VoiceFlagCount> mFlags{};
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uint mNumCallbackSamples{0};
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struct TargetData {
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int FilterType;
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al::span<FloatBufferLine> Buffer;
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};
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TargetData mDirect;
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std::array<TargetData,MAX_SENDS> mSend;
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/* The first MaxResamplerPadding/2 elements are the sample history from the
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* previous mix, with an additional MaxResamplerPadding/2 elements that are
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* now current (which may be overwritten if the buffer data is still
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* available).
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*/
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using HistoryLine = std::array<float,MaxResamplerPadding>;
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al::vector<HistoryLine,16> mPrevSamples{2};
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struct ChannelData {
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float mAmbiHFScale, mAmbiLFScale;
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BandSplitter mAmbiSplitter;
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DirectParams mDryParams;
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std::array<SendParams,MAX_SENDS> mWetParams;
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};
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al::vector<ChannelData> mChans{2};
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Voice() = default;
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~Voice() = default;
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Voice(const Voice&) = delete;
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Voice& operator=(const Voice&) = delete;
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void mix(const State vstate, ContextBase *Context, const uint SamplesToDo);
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void prepare(DeviceBase *device);
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static void InitMixer(al::optional<std::string> resampler);
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DEF_NEWDEL(Voice)
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};
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extern Resampler ResamplerDefault;
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#endif /* CORE_VOICE_H */
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